Adaptive noise cancelling microphone system

ABSTRACT

An adaptive noise canceling microphone system for extracting a desired signal, in particular a desired speech signal, comprising two microphones being arranged at a predefined distance from each other; a signal forming system (SFS) being adapted to receive a first and second input signals resulting from sounds received by the two microphones wherein an acoustical signal component in the first input signal is determined, wherein an acoustical signal component in the second input signal is determined, wherein the acoustical signal component in the first input signal is enhanced to generate a speech enhanced signal, and wherein the acoustical signal component in the second input signal is suppressed to generate a speech nulled signal; an adaptive noise cancellation filtering circuit being adapted to receive the speech enhanced signal and the speech nulled signal, wherein the noise in the speech enhanced signal is cancelled using the speech nulled signal as reference, thereby generating an output filtered signal representing the desired signal.

This invention relates to an adaptive noise canceling microphone systemwhich is capable of extracting desired signal from sounds containingnoise.

BACKGROUND OF THE INVENTION

A noise canceling microphone system is gaining more importance nowadays,especially with the development of multimedia applications and wirelesscommunication technologies. Although various solutions were proposed toenhance desired signal extraction, particularly desired speech, in noisyenvironments, there is still room for improvement in order to obtain ahigh Signal-to-Noise (SNR) ratio using very few microphones.

Various methods were commonly used to increase the SNR of desired speechsignal. In a known speech enhancement method, a single microphone isused to pick up the desired speech signal with noise. The noise spectrumis estimated and subtracted from the speech signal (containing thenoise) picked up by the microphone. In this way, the desired speechsignal is separated from the noise. However this method is onlyeffective with stationary noise, and also introduces high distortion tothe desired speech signal.

Another known noise cancellation method uses two microphones, with onemicrophone located near the source of the desired signal, and anothermicrophone located near the noise source. Thus, the signal picked up bythe microphone arranged near the noise source can be used to adaptivelycancel the noise signal from the signal picked up by the microphonearranged near the desired speech signal. This method is not practical inmost applications as it is very difficult to arrange a microphone nearthe noise source.

A further known microphone array system uses more than two microphones.The system uses a spatial and temporal filtering method to enhance thedesired speech signal from a specific direction and over an interestedfrequency band, and suppress signals from other directions. The desiredsignal obtained with this system has a high SNR. However the use of morethan two microphones results in a large system, unsuitable for use inmany mobile applications.

In the system disclosed in U.S. Pat. No. 4,742,548, more than threemicrophones are used to form a unidirectional microphone system fornoise cancellation. Since no adaptive signal processing method is usedin this system, the spatial response of this microphone system is fixed.This makes the whole system inflexible.

According to the system described in U.S. Pat. No. 5,226,076, twomicrophones are used to form a first-order microphone system for noisecancellation. However, this microphone system uses only the differentialproperty of sound field to form a fixed beam pattern, the performance ofthe system is therefore poor, especially in environments withcomplicated noise signals.

system disclosed in WO 0,195,666, a cardioid-type directional microphoneand an omni-directional microphone are combined in an acousticallycoupled way. The two microphones, together with an adaptive controlcircuit, produce a very narrow 3-dimensional beam for acquiring thedesired speech signal. However, the direction of the 3-dimension beam inthis system is fixed. In other words, when the desired signal is comingfrom another direction which is different from the direction the beam isprojected, the microphone system is not able to acquire the desiredsignal. The physical orientation of the system needs to be adjusted suchthat the desired signal falls into the direction of the 3-dimensionalbeam in order for the desired signal to be acquired. This makes the useof the system inconvenient.

Therefore, a noise cancellation microphone system with a high SNR andwhich is convenient to use is desired.

SUMMARY OF THE INVENTION

According to the invention, an adaptive noise canceling microphonesystem is provided, the adaptive noise canceling microphone systemcomprises two microphones being arranged at a predefined distance fromeach other, a signal forming system (SFS) for receiving a first andsecond input signal resulting from sounds received by the twomicrophones wherein an acoustical signal component in the first inputsignal is determined, wherein an acoustical signal component in thesecond input signal is determined, wherein the acoustical signalcomponent in the first input signal is enhanced to generate a speechenhanced signal and wherein the acoustical signal component in thesecond input signal is suppressed to generate a speech nulled signal,and an adaptive noise cancellation filtering circuit for receiving thespeech enhanced signal and the speech nulled signal, wherein the noisein the speech enhanced signal is cancelled using the speech nulledsignal as reference, thereby generating an output filtered signalrepresenting the desired signal.

The noise canceling microphone system according to the invention usestwo general microphones to form a flexible microphone system. The twomicrophones are arranged preferably between 1 cm (centimeter) and 10 cmfrom each other, and further preferably between 2 cm and 3 cm from eachother. It should however be noted that the microphones may be also beplaced. The microphones used may be any kind of microphones, includingomni-directional microphones or directional microphones. The microphonesgenerate signals corresponding to the sounds they received, and thesignals are received as first and second input signals by the SignalForming System (SFS).

The SFS processes the first and second input signals using filteringtechniques to generate the speech enhanced signal and the speech nulledsignal. The acoustical signal component in the first and the acousticalsignal component in the second input signal are an estimation of thedesired signal in the first input signal and the second input signal,respectively. Therefore, the speech enhanced signal represents thesounds with the desired signal enhanced, and the speech nulled signalrepresents the sounds with the desired signal suppressed.

The microphone system according to the invention does not specificallyrequire use of a combination of an omnidirectional microphone and adirectional microphone, as disclosed in WO 0,195,666, to form a3-dimensional beam for noise canceling. The beam is however formed bycontrolling the parameters of the filters used in the SFS. Moreover withthe SFS, the desired signal need not come from a pre-defined directionas the SFS is able to detect the desired signal and form the3-dimensional beam which is directed, automatically or manually, towardsthe direction of the desired signal. This thus allows the microphonesystem according to the invention to have a higher flexibility than themicrophone system described in WO 0,195,666.

Both the speech enhanced signal and the speech nulled signal generatedby the SFS are received by the adaptive noise cancellation filteringcircuit for canceling the noise from the speech enhanced signal, usingthe speech nulled signal as a reference, to generate the output filteredsignal which represents the desired signal. The noise cancellationprocess may be implemented using a Least Means Squared (LMS) filteringtechnique as disclosed in B. Widrow et al., “Adaptive NoiseCancellating: Principles and Applications,” in Proc. IEEE, 1975 or across-talk cancellation technique as disclosed in US 20010048740 A1.

To further reduce noise from the output filtered signal (representingthe desired signal), a post processing filtering circuit is provided.The post processing (PP) filtering circuit is adapted to receive theoutput filtered signal, with both the speech enhanced signal and thespeech nulled signal as reference, to reduce any noise present in theoutput filtered signal, and to generate a digital result signal. Thisdigital result signal represents the output filtered signal (which isalso the desired signal) with noise further reduced.

An adaptive noise canceling microphone (ANCM) controller is preferablyused to control at least one of the filters used in the SFS and theadaptive noise cancellation filtering circuit by updating thecoefficients of the filters. The ANCM controller is adapted to receivethe speech enhanced signal, the speech nulled signal and the outputfiltered signal, and generate an SFS control and an adaptive noisecancellation filter (ANCF) control signal to update the coefficients ofthe filters used in the SFS and the adaptive noise cancellationfiltering circuit, respectively. In the preferred embodiment of theinvention, the ANCM controller is also adapted to generate a PP controlsignal for updating the coefficients of at least one filter used in thePP filtering circuit.

The ANCM controller processes the speech enhanced signal, the speechnulled signal and the output filtered signal to determinecharacteristics of the input signals, and generates the optimizedcontrol signals for updating the coefficients of the filters used in theSFS, the adaptive noise cancellation filtering circuit and the PPfiltering circuit. Specifically, each control signal represents a stepsize value for the respective filters, wherein the step size value is aparameter used for updating the weights or coefficients of the filtersin this way, the updating of the weights or coefficients of therespective filters can be performed more efficiently.

The step size value used in the filters is important for ensuring thefilters to work stably and effectively. The step size value ispreferably set to a small value when a desired signal, in particular adesired speech, is detected.

The step size value is preferably set to a large value when the desiredspeech is not detected. When no signal is detected (when there issilence), the step size value is preferably set to zero resulting in theweight of the filter remaining unchanged, so that the filters canoperate properly and stably.

Since the ANCM controller controls the step size values used forupdating the filters in such a way that the values change in an adaptivemanner depending on characteristics of the desired signal and are notassigned a fixed value according to the state of the art, it allows themicrophone system according to the invention to be able to effectivelycancel unwanted noise and remain the desired signal from the inputsignals.

The SFS according to the invention comprises a first Finite ImpulseResponse (FIR) filter, a second Finite Impulse Response (FIR) filter, afirst adder and a second adder. The first FIR filter and the second FIRfilter receive the first input signal and are adapted to determine theacoustical signal component in the first input signal, and to generate afirst and second FIR filtered signal, respectively, which represent theacoustical signal component in the first input signal. The first adderadds the first filtered signal to the second input signal and generatesthe speech enhanced signal. The second adder subtracts the second FIRfiltered signal from the second input signal and generates the speechnulled signal.

The coefficients of the FIR filters are designed to form adirectionality pattern, in particular a 3-dimensional beam, directed atthe desired signal such that any signals within this beam are retainedas desired signals and any signals outside this beam are rejected asnoise. The speech enhanced signal containing the desired signalextracted from these FIR filters however has a low SNR and therefore,further processing is thus needed to obtain the desired signal with ahigh SNR.

The coefficients of the FIR filters are preferably calculated by a LeastMeans Squared (LMS) algorithm block. In this case, the LMS algorithmblock is adapted to receive the first input signal, the speech nulledsignal and the SFS control signal to generate a LMS signal representingthe coefficients of the FIR filters. The LMS algorithm block is able tochange the coefficients of the FIR filters adaptively such that thedirectionality pattern is always directed at the desired signal.

In an alternative embodiment of the invention, information representingthe coefficients of the first and second FIR filter and the second FIRfilter is stored in a first Read Only Memory (ROM) and a second ROM,respectively. The coefficient information which is stored in the ROMs ispre-calculated by experiment with only the desired signal (without anynoise or interference). The stored coefficient information is then usedas the coefficients for the filters according to user's preference.

The adaptive noise cancellation filtering circuit according to theinvention can be implemented in an LMS configuration which comprises anadaptive filter and an adder. The adaptive filter is adapted to receivethe speech nulled signal, the ANCF control signal and an error signal(or the output filtered signal) to generate a filtered signal. The ANCFcontrol signal is used to control the step size for updating thecoefficients of the adaptive filter, and the error signal providesfeedback information to the adaptive filter. The adder subtracts thefiltered signal from the speech enhanced signal, and generates the errorsignal. A delay circuit is preferably added before the adder, whereinthe delay circuit is adapted to receive and delay the speech enhancedsignal and to generate a delayed signal.

In the preferred embodiment of the invention, an additional adaptivefilter and an additional adder are used to form a cross-talkconfiguration. The additional adder receives an additional filteredsignal and subtracts it from the speech nulled signal, and generates anadditional error signal which is then received by the adaptive filter.The additional adaptive filter is adapted to receive the error signal,the additional error signal and the ANCF control signal and to generatethe additional filtered signal which is also received by the additionaladder. The addition filtered signal represents an estimated component ofthe desired signal in the error signal, the ANCF control signal is usedto update the coefficients of the additional adaptive filter, and theadditional error signal is used as a feedback information to theadditional adaptive filter. This cross-talk configuration isadvantageous because the filtered signal from the adaptive filter is amore accurate estimation of the noise signal than the filtered signal inthe LMS configuration described earlier. Therefore, the output filteredsignal from the adder in this cross-talk configuration represents thedesired signal with a higher SNR.

The PP filtering circuit according to the invention comprises a FIRfilter and a coefficient calculation block. The FIR filter of the PPfiltering circuit is adapted to receive the output filtered signal andto generate the digital result signal. The coefficients of the FIRfilter are updated by the coefficient calculation block. The coefficientcalculation block is adapted to receive the speech enhanced signal, thespeech nulled signal and the output filtered signal and to generate acoefficient signal used for updating the coefficients of the FIR filter.In the preferred embodiment of the invention, the coefficientcalculation block is adapted to also receive the PP control signal as aninput for generating the coefficient signal. The PP control signal isable to provide a suitable coefficient parameter, in particular a stepsize value, for updating the coefficients of the FIR filter, thusresulting in most optimal coefficients or weights for the FIR filter.

The adaptive noise cancellation controller according to the inventioncomprises a power estimator unit for each of the first input signal, thesecond input signal and the output filtered signal, a silence detectorunit, a signal detector unit, an ANCF controller unit and a SFScontroller unit.

The power estimator unit is adapted to generate a power signalcorresponding to the estimated power of the received signal. The outputpower of the power estimation unit (i.e. power signal) is designed tofollow any rapid changes in its input. When the desired signal containedin its input fades away, the power signal should decay slowly so thatthe weight of the adaptive filter of the adaptive noise cancellationfiltering circuit is not updated wrongly when the desired signal ispresent.

The silence detector unit detects whether the sounds received by themicrophones correspond to a “silent” signal, and is adapted to generatethe first output signal accordingly. In other words, the silencedetector unit detects the case when there is no strong background noise,interference signal or any desired signal and generates the first outputsignal indicating when such mentioned noise or signal are not present.When a “silent” signal is indicated by the first output signal, it meansthat only random circuit noise such as channel noise is present.

The signal detector unit detects whether the desired signal, inparticular desired speech, is present, and is adapted to generate thesecond output signal accordingly. The second output signal providesinformation on whether the desired speech or interference noise, or acombination of both the desired speech and interference noise arepresent.

The ANCF controller unit is adapted to receive the first output signaland the second output signal to determine the characteristic of theinput signals, and to use the information thereof to determine anappropriate step size value to be used for updating the weights orcoefficients of the adaptive filter(s) used in the adaptive noisecancellation filtering circuit. The ANCF controller unit is adapted togenerate the ANCF control signal which corresponds to the appropriatestep size value.

Similarly, the SFS controller unit is adapted to receive the firstoutput signal and the second output signal to determine thecharacteristic of the input signals and to use the information thereofto determine an appropriate step size value to be used for updating theweights or coefficients of the FIR filters of the SFS. The SFScontroller unit is adapted to generate the SFS control signal whichcorresponds to the appropriate step size value.

In the preferred embodiment of the invention, the adaptive noisecancellation controller unit further comprises a PP controller unitwhich is adapted to receive the first and second output signal todetermine the characteristic of the input signals and to use theinformation thereof to determine an appropriate step size value forupdating the weights or coefficients of the FIR filter of the PPfiltering circuit.

The ANCF controller unit, the SFS controller unit and the PP controllerunit use a counter based method combined with an adaptive update methodto get a stable control on the step size values of the filters.Therefore, a smooth control signal from the controllers is ensured evenunder uncertain system perturbation. This prevents the adaptive noisecanceling microphone system from becoming unstable, and short uncertainperturbation will not affect the performance of the system. Furthermore,since the step size values of the filters are not fixed but change in anadaptive manner, the weights of the filters are always obtained at theoptimal value even when the input signals change.

According to the invention, a first amplifier and a second amplifier areused to amplify the signals resulting from the sounds picked up by themicrophones. The amplified signals are filtered by a first low passfilter and a second low pass filter to generate analog amplified andfiltered signals. The amplified and filtered signals are converted todigital signals using a first Analog-to-Digital (A/D) Converter and asecond Analog-to-Digital (A/D) converter, the digital signals are thefirst and second input signals which are inputs to the SFS.

The output of the PP filtering circuit is the digital result signal,which is in digital form. To use the digital result output in mostanalog devices, such as speakers, an Analog-to-Digital (A/D) converteris used to convert the digital result signal into an analog signal,thereby generating a result signal output. The result signal output,which is in analog form, can then be received by an analog input device,for example, speaker.

Other objects, features and advantages according to the invention willbe presented in the following detailed description of the illustratedembodiments when read in conjunction with the accompanying drawings.

BRIEF DESCRIPTION OF THE DRAWINGS

FIG. 1 shows the block diagram of an adaptive noise cancellationmicrophone system, comprising a processing unit according to theinvention.

FIG. 2 shows the block diagram of the processing unit shown in FIG. 1according to the invention.

FIG. 3 shows the detail block diagram of the adaptive noise cancelingsystem according to the invention.

FIG. 4 shows the detail block diagram of the signal forming systemaccording to the invention.

FIG. 5 shows the detail block diagram of the adaptive noise cancellationfiltering circuit according to the invention.

FIG. 6 shows the detail block diagram of the adaptive noise cancellationfiltering circuit according to the preferred embodiment of theinvention.

FIG. 7 shows the detail block diagram of the post processing filteringcircuit according to the preferred embodiment of the invention.

FIG. 8 shows the detail block diagram of the coefficient calculationblock of the post processing filtering circuit according to thepreferred embodiment of the invention.

FIG. 9 shows the detail block diagram of the adaptive noise cancellationcontroller according to the preferred embodiment of the invention.

FIG. 10 shows the flow diagram of a power estimator unit according tothe invention.

FIG. 11 shows the flow diagram of the silence detector unit according tothe invention.

FIG. 12 shows the flow diagram of the signal detector unit according tothe invention.

FIG. 13 shows the flow diagram of the signal forming system controllerunit according to the invention.

FIG. 14 shows the flow diagram of the adaptive noise cancellation filtercontroller unit according to the invention.

FIG. 15 shows the flow diagram of the post processing filter controllerunit according to the preferred embodiment of the invention.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS OF THE INVENTION

FIG. 1 shows the block diagram of the adaptive noise cancelingmicrophone system 10 according to the invention. The system 10 comprisesa first microphone 11, a second microphone 12 and a processing unit 15.The processing unit 15 receives signals 13,14 resulting from soundsreceived by the microphones 11,12. An output result signal 16 isgenerated by the processing unit 15.

The two microphones 11,12 are arranged at a predefined distance,preferably between 2 cm and 3 cm, from each other so as to form a3-dimensional beam for receiving sounds using digital signal processingtechniques. The forming of the 3-dimensional beam is controlled by asignal forming system (described later) in the processing unit 15.

A detailed block diagram of the processing unit 15 is shown in FIG. 2.The signals 13,14 generated by the microphones 11,12 are amplified in afirst amplifier 20 and a second amplifier 21, respectively. Theamplified signals are filtered in a first and second low pass filter22,23 to remove any high frequency components. The amplified andfiltered signals are subsequently converted to digital signals 26,27using a first and second Analog-to-Digital (A/D) converters 24,25 to beprocessed by an adaptive noise canceling system 28.

The digital signals 26,27 are received as first and second input signals26,27 by the adaptive noise canceling system 28 for processing to remainthe desired signal and cancel the noise or interference signals. Theoutput of the adaptive noise canceling system 28 is a digital resultsignal 29. The digital result signal 29 can be further processed byother systems, or converted to an analog signal for driving outputdevices like speakers. According to the invention, the digital resultsignal 29 is converted to an analog signal, known as the result signaloutput 16, using a Digital-to-Analog (A/D) converter 30.

FIG. 3 shows the block diagram of the adaptive noise canceling system 28according to the preferred embodiment of the invention. The adaptivenoise canceling system 28 comprises of the signal forming system (SFS)31, an adaptive noise cancellation filtering circuit 32, a postprocessing (PP) filtering circuit 33 and an adaptive noise cancellationmicrophone (ANCM) controller 34.

The SFS 31 is adapted to receive the first input signal 26, the secondinput signals 27 and a SFS control signal 38 to form a speech enhancedsignal 35 and a speech nulled signal 36. The speech enhanced signal 35represents sounds received with the desired signal enhanced, whereas thespeech nulled signal 36 represents the sounds received with the desiredsignal suppressed. In other words, the speech nulled signal 36represents noise or interference signal.

The adaptive noise cancellation filtering circuit 32 is adapted toreceive the speech enhanced signal 35, the speech nulled signal 36 andan adaptive noise canceling filter (ANCF) control signal 39, and togenerate an output filtered signal 37. The PP filtering circuit 33 isadapted to receive the speech enhanced signal 35, the speech nulledsignal 36, the output filtered signal 37 and a PP control signal 40, andto generate the digital result signal 29.

The adaptive noise cancellation controller 33 is adapted to receive thespeech enhanced signal 35, the speech nulled signal 36 and the outputfiltered signal 37, and to generate the SFS control signal 39, the ANCFcontrol signal 38 and the PP control signal 40 for updating coefficientsof at least one filters used in the SFS 31, the adaptive noisecancellation filtering circuit 32 and the post processing filteringcircuit 33, respectively.

The ANCM controller 34 updates the coefficients of the at least onefilters in the SFS 31, the adaptive noise cancellation filtering circuit32 and the post processing filtering circuit 33 by controlling the stepsize value of the filters used in them.

The detail implementation of the SFS 31 is shown in FIG. 4. The SFS 31comprises a first and second finite impulse response (FIR) filter 41,42,and a first and second adder 43,44. The first and second FIR filters41,42 are adapted to receive the first input signal 26 and to generate afirst and second FIR filtered signal 48,49, respectively. The first andsecond FIR filters 41,42 are used to approximate the desired signalcomponents in the first input signal 26 which corresponds to the desiredsignal components in second input signal 27, resulting the first andsecond filtered signals 48,49.

The first adder 43 adds the first filtered signal 48 to the second inputsignal 27 to produce the speech enhanced signal 35, which is the soundsreceived with the desired signal enhanced. The second filtered signal 49is subtracted from the second input signal 27 by the second adder 44 toform the speech nulled signal 36.

In the preferred embodiment of the invention, the coefficients of thefirst and second FIR filters 41,42 are updated by a Least Means Squared(LMS) algorithm block 45. The LMS algorithm block 45 is adapted toreceive the first input signal 26, with the SFS control signal 38 andthe speech nulled signal 36 as reference, to update coefficients of theFIR filters 41,42. The coefficients of the FIR filters 41,42 are updatedusing the following equation: $\begin{matrix}{{W( {n + 1} )} = {{W(n)} + \frac{{m_{sfs}(n)}{X(n)}{e(n)}}{{{X(n)}}^{2}}}} & (1)\end{matrix}$wherein

-   W(n) is the weight of the FIR filters 41,42,-   m_(sfs)(n) is the step size value of the FIR filter 41,42,-   X(n) is the input signal 26, and-   e(n) is the output of the second adder 44 or the speech nulled    signal 36.

The weights or coefficients of the FIR filters 41,42 change adaptivelyaccording to the characteristic of the desired signal contained in theinput signals 26,27.

The coefficients for the FIR filters 41,42 may be pre-calculated, andsuch pre-calculated coefficient information may be stored in a first andsecond Read Only Memory (ROM) 46,47 in an alternative embodiment. Theinformation stored in the first and second ROM 46,47 can then be loadedand used as the weights or coefficients for the first and second FIRfilters 41,42, respectively, according to the user's preference. In sucha case, the LMS block 45 is no longer needed.

FIG. 5 shows an implementation of the adaptive noise cancellationfiltering circuit 32 according to the invention. The adaptive noisecancellation filtering circuit 32 comprises a delay circuit 51, an adder52 and an adaptive filter 53. The speech enhanced signal 35 is delayed apredetermined number of samples by a delay circuit 51 to generated adelayed signal 54. The adaptive filter 53 is adapted to receive andestimate the component of noise based on the speech nulled signal 36 andto generate a filtered signal 55 representing the noise components. Thefiltered signal 55 is subtracted from the delayed signal 54 by the adder52 to generate an error signal 37 which is also the output filteredsignal 37.

The adaptive filter 53 is also adapted to receive the error signal 37 asfeedback information, and the ANCF control signal 39 for controlling thestep size value for proper updating of its filter coefficients.

FIG. 6 shows the implementation of the adaptive noise cancellationfiltering circuit 32 using a cross-talk configuration according to thepreferred embodiment of the invention. The cross-talk configurationcomprises of the adaptive filter 53, the adder 52, an additionaladaptive filter 56 and an additional adder 57. The adder receives thespeech enhanced signal 35 and the filtered signal 55 to generate theoutput filtered signal or the error signal 37. The additional adder 57is adapted to receive the speech nulled signal 36 and an additionalfiltered signal 59, and to generate an additional error signal 58. Theadditional filtered signal 59 is generated by the additional adaptivefilter 56. The additional adaptive filter 56 is also adapted to receivethe error signal 37 as input and the additional error signal 58 asfeedback information. Similarly, the adaptive filter 53 is adapted toreceive the additional error signal 58 as input and the error signal 37as feedback information. The step size value for updating thecoefficients of both the adaptive filters 53,56 are provided by the ANCFcontrol signal 39.

FIG. 7 shows the detailed block diagram of the PP filtering circuit 33.The PP filtering circuit 33 comprises a FIR filter 71 and a coefficientcalculation block 72. The FIR filter 71 is adapted to receive the outputfiltered signal 37 to further reduce any noise present in the outputfiltered signal 37, and to generate the digital result signal 29. Thecoefficients for the FIR filter 71 are updated by the coefficientcalculation block 72. The coefficient calculation block 72 is adapted toreceive the speech enhanced signal 35, the speech nulled signal 36 andthe output filtered signal 37 as references, together with the PPcontrol signal 40 as the step size value, and to generate a coefficientsignal 73 representing the coefficients of the FIR filter 71. Thedetailed implementation of the coefficient calculation block 72 can besummarized in FIG. 8.

The speech enhanced signal 35, speech nulled signal 36 and outputfiltered signal 37 are transformed by each separate Fast FourierTransform (FFT) unit 81,82,83 into their respective frequency domain,represented by m1(k), m2(n) and e(k), respectively. The signals m1(k)and m2(n) are received by two separate spectral power estimation units84,85 for generating corresponding power signals P_(m1)(k) and P_(m2)(k)using the following equations:P _(m1)(k)=aP _(m1)(k−1)+(1−a).m 1(k)×m 1·(k)  (2) P _(m2)(k)=aP _(m2)(k−1)+(1−a).m 2(k)×m 2·(k)  (3)wherein a is the forgetting factor for the power computation, and theoperator * denotes the complex conjugate.

The signals m1(k) and e(k) are also used to estimate thecross-correlation function P_(m1e)(k) in the correlation estimation unit86 using the following equations:P _(m1e)(k)=aP _(m1e)(k−1)+(1−a).m 1(k)×e·(k)  (4)

The average correlation function AP_(m1e)(k) is further obtained in anaverager unit 87 using the following equation: $\begin{matrix}{{A\quad{P_{m\quad l\quad e}(j)}} = {\sum\frac{P_{m\quad e}(k)}{L}}} & (5)\end{matrix}$wherein for all k in the FFT block, and L is the length of the FFTblock. The signal AP_(m1e)(k) is then used to generated a weight signalA_(me)(j) in a weighted estimation unit 88: $\begin{matrix}{{A_{m\quad e}(j)} = \frac{a}{( {{A\quad{P_{m\quad l\quad e}(j)}} + b} )^{c}}} & (6)\end{matrix}$wherein a, b and c are predefined positive constants. A post processingresponse calculator (PPRC) 89 is used to determine the coefficients forthe FIR filter 71 using the following equation: $\begin{matrix}{{F(k)} = \frac{P_{m\quad l\quad e}(k)}{( {{P_{mle}(k)} + {{A_{m\quad e}(j)} \times {P_{m\quad 2}(k)}}} )}} & (7)\end{matrix}$which is subsequently transformed back to the time domain by an InversedFFT unit 90 to obtain the coefficient signal 73 for the FIR filter 71.

FIG. 9 shows the block diagram of the ANCM controller 34 according tothe preferred embodiment of the invention. The ANCM controller 34comprises three power estimator units 91,92,93, a silence detector unit97, a signal detector unit 98, an adaptive noise canceling filter (ANCF)controller unit 101, a SFS controller unit 102 and a PP filtercontroller unit 103.

The three power estimator units 91,92,93 are adapted to receive and toestimate the power of the three signals 35,36,37. The outputs of thepower estimator are three power signals 94,95,96. The power signals94,95,96 are received as inputs by both the silence detector unit 97 andthe signal detector unit 98. The silence detector unit 97 is adapted togenerate a first output signal 99, and the signal detector unit 98 isadapted to generate a second output signal 100. The ANCF controller unit101 is adapted to receive both the first and second output signal 99,100and to generate the ANCF control signal 39. Similarly, the SFScontroller unit 102 and the PP filter controller unit 103 are adapted toreceive the first and second output signal 99,100 and to generate theSFS control signal 38 and the PP control signal 40, respectively.

The ANCF control signal 39, the SFS control signal 38 and the PP controlsignal 40 are used to update the step size value, and hence thecoefficients of the filters used in the SFS 31, the adaptive noisecanceling filtering circuit 32 and PP filtering circuit 33,respectively.

The detail implementation of each of the unit of the ANCM controller isillustrative from FIG. 10 to FIG. 15.

FIG. 10 shows a flow chart of one of the power estimator units 91,92,93.At an initialization step 201, the initial power of the signals 35,36,37are initialized to zero:p ₁(n),p ₂(n),p ₃(n)=0  (8)wherein p₁(n), p₂(n) and p₃(n) are the power of the signals 35,36,37,respectively, corresponding to the power signals 94,95,96.

Similarly, the initial power update forgetting factors of the powerestimators 91,92,93 are also initialized to some nonzero value, such as0.5:a ₁(n),a ₂(n),a ₃(n)=0.5  (9)wherein a₁(n), a₂(n) and a₃(n) are the power update forgetting factorsof power estimators 91,92,93, respectively. The minimum value of thepower update forgetting factors is a_(min) and the maximum value isa_(max).

After the initialization step 201, the signals 35,36,37 are collected atstep 202. The power of each of the input signals 35,36,37 are thenupdated in step 203 using the following formula:p(n)=(1−a(n))*p(n−1)+a(n)*|x(n)|²  (10)wherein x(n) is any of the signals 35,36,37, p(n) is the current powerof any of the input signals p₁(n), p₂(n) or p₃(n), and p(n−1) is theprevious power.

In step 204, the value of the current power P(n) is compared with avalue of the previous power P(n−1) before updating to determine if thepower P(n) of the corresponding input signal is stationary. Thecomparison of the current power P(n) and previous power P(n−1) isperformed using the following formulae:D _(p) =|p(n)−p(n−1)  (11)D_(p)<T₁  (12)wherein T₁ is a parameter which is used to determine whether theestimated signal power is stable. When D_(p) is smaller than T₁, itmeans that the power P(n) of the input signal is substantiallystationary. In this case, the power P(n) is generated as the outputpower of the signal in step 205.

When D_(p) is greater than T₁, it means that the power of the inputsignal has large changes between the current power P(n) and previouspower P(n−1), and the power estimator unit then compares the powervalues in step 206, using the following:p(n)>p(n−1)  (13)If p(n) is greater than p(n−1), the power estimator unit updates thepower updating factor a(n) in step 207 using the following formula:a(n)=a(n−1)+D ₁  (14)wherein a(n) is the current power updating factor a₁(n), a₂(n), a₃(n) ofany of the power estimator unit 91,92,93, a(n−1) is the previous powerupdating factor, and D₁ is a step size parameter for updating the powerupdating factor a(n).

The updated power updating factor a (n) is further compared in step 208to determine if it exceeds a maximum value:a(n)<a _(max)  (15)wherein a_(max) is the maximum allowable value of a (n). If a(n) issmaller than a_(max), the power p(n) is generated as the output power instep 205. If a (n) is greater than a_(max), then a(n) is assigned thevalue of a_(max) in step 209 and the power p(n) is generated as theoutput power in step 205.

If p(n) is determined to be smaller than p(n−1) during comparison instep 206, the power estimator unit updates the power updating factora(n) in step 210 using the following formula:a(n)=a(n−1)−D ₂  (16)and determines in step 211 if a(n) is greater a_(min):a(n)>a _(min)  (17)wherein D₂ is another step size parameter for updating the powerupdating factor a(n), and a_(min) is a minimum allowable value of a(n).If a(n) is greater than a_(min), the power p(n) is generated as theoutput power in step 205. If a(n) is smaller than a_(min), then a(n) isassigned the value of a_(min) in step 212 and the power p(n) isgenerated in step 205.

The power p(n) generated from step 205 is used as the output power forthe silence detector unit 97 and the signal detector unit 98. Thesilence detector unit 97 receives the power signals p₁(n), p₂(n) andp₃(n) 94,95,96 as inputs from the respective power estimator units91,92,93 in step 300 of the flow chart shown in FIG. 11.

In step 301, the power signal p₃(n) 96 from the power estimator unit 93corresponding to the input signal 37 is compared with the noise powerp_(n)(n):p ₃(n)>k ₁ .p _(n)(n)  (18)wherein k₁ is a threshold value used to detect “silence”. If p₃(n) isgreater than k₁.p_(n)(n), the “silence” is set to “0” in step 303,indicating that “silence” is not detected, and interference signaland/or desired signal is present.

If p₃(n) is smaller than k₁.p_(n)(n), the silence detector unit checksif both p₁(n) and p₂(n) are greater than k₁.p_(n)(n) in step 302:p ₁(n)>k ₁ .p _(n)(n)  (19)p ₂(n)>k ₁ .p _(n)(n)  (20)

If both p₁(n) and p₂(n) are greater than k₁.p_(n)(n), then “silence” isset to “0” in step 303, else it checks if both p₁(n) and p₂(n) aresmaller than k₂.p_(n)(n) in step 304:p ₁(n)<k ₂ .p _(n)(n)  (21)p ₂(n)<k ₂ .p _(n)(n)  (22)wherein k₂ is another threshold value used to detect “silence”. If bothp₁(n) and p₂(n) are smaller than k₂.p_(n)(n), then the noise powerp_(n)(n) is updated in step 305 using the following formula:$\begin{matrix}{{p_{n}(n)} = {{d\quad{p_{n}( {n - 1} )}} + \frac{( {1 - d} ) \cdot ( {{p_{1}(n)} + {p_{2}(n)}} )}{2}}} & (23)\end{matrix}$wherein d is a fixed parameter which can be selected as 0.9999 forsmoothing the background power value. The “silence” is set to “1”,indicating that there are no signal present. If in step 304, either oneor both of p₁(n) and p₂(n) are greater than k₂.p_(n)(n), the “silence”is set to “1” directly in step 306 without updating the value of thenoise power p_(n)(n).

FIG. 12 shows a flow chart of the signal detector unit 98 for alsoreceiving the power signals p₁(n), p₂(n) and p₃(n) 94,95,96 as inputsfrom the respective power estimator units 91,92,93 in step 400.

In step 401, the power signal p₃(n) is compared with p₁(n):p ₃(n)>k ₃ .p _(n)(n)  (24)wherein k₃ is a first threshold value for detecting the desired signal.If p_(3(n)) is smaller than k₃.p₁(n), then the signal detector unitdetermines in step 402 if:p ₁(n)>k ₅ .p ₂(n)  (25)wherein k₅ is a second threshold value for detecting desired signal. Ifp₁(n) is greater than k₅.p₂(n), the “signal” is set to “1” in step 403,indicating that desired signal is present. If p₁(n) is smaller thank₅.p₂(n), the “signal” is set to “0” in step 404, indicating thatdesired signal is not present.

If in step 401, p_(3(n)) is greater than k₃.p₁(n), then the signaldetector unit determines further in step 405 if:|p ₁(n)−p ₂(n)|<k ₄ .p ₁(n)  (26)wherein k₄ is a third threshold value for detecting the desired signal.If |p₁(n)−p₂(n)| is smaller than k₄.p₁(n), the ANCM controller 34returns to the initialization state in step 201, and all systemvariables are initialized to the initial values. Otherwise thecontroller 34 proceeds to step 402.

The output of the silence and signal detector units 97,98 are the valuesof “silence” and “signal”, respectively. Knowing the values of “silence”and “signal” enables one to determine if the signal contained by thesounds received comprises of desired signal, interference signal ornoise, or a combination of them.

The outputs from both the silence detector unit 97 and the signaldetector unit 98 are received by the ANCF controller 39, the SFScontroller 38 and the PP filter controller 40.

The flow chart of the SFS controller 38 is shown in FIG. 13. In step500, the outputs from the silence and signal detector units 97,98 arereceived by the SFS controller 38. In step 501, the value of “silence”is determined if it is set at “1”. If “silence” is not set at “1”, i.e.set at “0”, then T_(sc) is initialized to 0 in step 502:T_(sfs-sc)=0  (27)wherein T_(sfs-sc) is a counter for determining “silence” in the SFScontroller 38.

In step 503, the value of “speech” is determined if it is set at “1”. If“speech” is not set at “1”, then T_(sfs-sp) is to 0 in step 504:T_(sfs-sp)=0  (28)wherein T_(sfs-sp) is a counter for determining “speech” in the SFScontroller 38. The step size parameter for the FIR filter of the SFSgenerated by the SFS controller 38 is the updated in step 505 using thefollowing formula:m _(sfs)(n)=(1−q).m _(sfs)(n−1)+q  (29)wherein m_(sfs)(n) is the step size parameter.

If “silence” is detected to be set at “1” in step 501, then T_(sfs-sc)is increment by 1 in step 506:T_(sfs-sc) =T _(sfs-sc)+1  (30)and the incremented value of T_(sfs-sc) is compared to see if it exceedsthe maximum value T_(sfs-sc) _(—) _(max) in step 507:T_(sfs-sc)>T_(sfs-sc) _(—) _(max).  (31)

If T_(sfs-sc) is greater than T_(sfs-sc) _(—) _(max), T_(sfs-sc) is setto T_(sfs-sc) _(—) _(max), and the output of the SFS controller 38,m_(sfs)(n), is set to “0” in step 508. Else, the SFS controller 38continues to determine the value of “speech” in step 503.

When the “speech” is set “1”, T_(sfs-sp) is increment by 1 in step 509:T _(sfs-sp) =T _(sfs-sp)+1  (32)and the incremented value of T_(sfs-sp) is compared to see if it exceedsthe maximum value T_(sfs-sp) _(—) _(max) in step 510:T_(sp)>T_(sp) _(—) _(max).  (33)

If T_(sfs-sp) is greater than T_(sfs-sp) _(—) _(max), T_(sfs-sp) is setto T_(sfs-sp) _(—) _(max), and the output of the SFS controller 38,m_(sfs)(n), is set to “0” in step 511. Else, the output, m_(sfs)(n),remains unchanged.

The flow chart of the ANCF controller 39 is shown in FIG. 14.

In step 600, the output from the silence and signal detector units 97,98are received by the ANCF controller 39. In step 601, the value of“silence” is determined if it is set at “1”. If “silence” is not set at“1”, i.e. set at “0”, then T_(sc) is initialized to 0 in step 602:T_(sc)=0  (34)wherein T_(sc) is a counter for determining “silence” in the ANCFcontroller 39.

In step 603, the value of “speech” is determined if it is set at “1”. If“speech” is not set at “1”, then T_(sp) is to 0 in step 604:

 T_(sp)=0  (35)

wherein T_(sp) is a counter for determining “speech” in the ANCFcontroller 39. The step size parameter for the adaptive filter of theadaptive noise cancellation filtering circuit 32 generated by the ANCFcontroller 39 is the updated in step 605 using the following formula:m _(ncf)(n)=(1−q).m _(ncf)(n−1)+q  (36)wherein m_(ncf)(n) is the step size parameter.

If “silence” is detected to be set at “1” in step 601, then T_(sc) isincrement by 1 in step 606:T _(sc) =T _(sc)+1  (37)and the incremented value of T_(sc) is compared to see if it exceeds themaximum value T_(sc) _(—) _(max) in step 607:T_(sc)>T_(sc) _(—) _(max).  (38)

If T_(sc) is greater than T_(sc) _(—) _(max), T_(sc) is set to T_(sc)_(—) _(max), and the output of the ANCF controller 39, m_(ncf)(n), isset to “0” in step 608. Else, the ANCF controller 39 continues todetermine the value of “speech” in step 603.

When the “speech” is set “1”, T_(sp) is increment by 1 in step 609:

 T _(sp) =T _(sp)+1  (39)

and the incremented value of T_(sp) is compared to see if it exceeds themaximum value T_(sp) _(—) _(max) in step 610:T_(sp)>T_(sp) _(—) _(max).  (40)

If T_(sp) is greater than T_(sp) _(—) _(max), T_(sp) is set to T_(sp)_(—) _(max), and the output of the ANCF controller 39, m_(ncf)(n), isset to “0” in step 611. Else, the output, m_(ncf)(n), remains unchanged.

The flow chart of the PP filter controller 40 is shown in FIG. 15.Similarly, the output from the silence and signal detector units 97,98are received by the PP filter controller 40 in step 700. In step 701,the value of “silence” is determined if it is set at “1”. If “silence”is not set at “1”, i.e. set at “0”, then T_(scp) is initialized to “0”in step 702:T_(scp)=0  (41)wherein T_(scp) is a counter for determining “silence” in the PP filtercontroller 40.

In step 703, the value of “speech” is determined if it is set at “1”. If“speech” is not set at “1”, then T_(spp) is to “0” in step 704:T_(spp)=0  (42)wherein T_(spp) is a counter for determining “speech” in the PP filtercontroller 40. The step size parameter for a noise canceling filtergenerated by the PP filter controller 40 is the updated in step 705using the following formula:b _(ncf)(n)=(1−j).b _(ncf)(n−1)+j  (43)wherein b_(ncf)(n) is the step size parameter.

If “silence” is detected to be set at “1” in step 701, then T_(scp) isincrement by 1 in step 706:T _(scp) =T _(scp)+1  (44)and the incremented value of T_(scp) is compared to see if it exceedsthe maximum value T_(scp) _(—) _(max) in step 507:T_(scp)>T_(scp) _(—) _(max).  (45)

If T_(scp) is greater than T_(scp) _(—) _(max), T_(scp) is set toT_(scp) _(—) _(max), and the output of the PP filter controller 40,b_(ncf)(n), is set to “0” in step 708. Else, the PP filter controller 40continues to determine the value of “speech” in step 703.

When the “speech” is set “1”, T_(spp) is increment by 1 in step 709:T _(spp) =T _(spp)+1  (46)

sand the incremented value of T_(spp) is compared to see if it exceedsthe maximum value T_(spp) _(—) _(max) in step 710:

 T_(spp)>T_(spp) _(—) _(max).  (47)

If T_(spp) is greater than T_(spp) _(—) _(max), T_(spp) is set toT_(spp) _(—) _(max), and the output of the PP filter controller 40,b_(ncf)(n), is set to “0” in step 711. Else, the output, b_(ncf)(n),remains unchanged.

While the embodiments of the invention have been described, they aremerely illustrative of the principles of the invention. Otherembodiments and configurations may be devised without departing from thespirit of the invention and the scope of the appended claims.

1. An adaptive noise canceling microphone system for extracting adesired signal, in particular a desired speech signal, comprising: twomicrophones being arranged at a predefined distance from each other; asignal forming system (SFS) being adapted to receive a first and secondinput signal resulting from sounds received by the two microphones,wherein an acoustical signal component in the first input signal isdetermined, wherein an acoustical signal component in the second inputsignal is determined, wherein the acoustical signal component in thefirst input signal is enhanced to generate a speech enhanced signal, andwherein the acoustical signal component in the second input signal issuppressed to generate a speech nulled signal; an adaptive noisecancellation filtering circuit being adapted to receive the speechenhanced signal and the speech nulled signal, wherein the noise in thespeech enhanced signal is cancelled using the speech nulled signal asreference, thereby generating an output filtered signal representing thedesired signal.
 2. The adaptive noise canceling microphone systemaccording to claim 1, further comprising a post processing filteringcircuit for reducing noise from the output filtered signal, wherein thepost processing filtering circuit is adapted to receive the outputfiltered signal, using the speech enhanced signal and the speech nulledsignal as reference, and to generate a digital result signalrepresenting the output filtered signal with noise reduced.
 3. Theadaptive noise canceling microphone system according to claim 2, furthercomprising an adaptive noise cancellation microphone (ANCM) controllerfor receiving the speech enhanced signal, the speech nulled signal andis the output filtered signal, the adaptive noise cancellationmicrophone (ANCM) controller being adapted to generate an adaptive noisecancellation filter (ANCF) control signal and an SFS control signal forupdating coefficients of at least one filter of the SFS and adaptivenoise cancellation filtering circuit, respectively, using the receivedspeech enhanced signal, the speech milled signal and the output filteredsignal.
 4. The adaptive noise canceling microphone system according toclaim 3, wherein the adaptive noise cancellation microphone controlleris adapted to generate a post processing filter control signal forupdating the coefficients of the at least one filter of the postprocessing filtering circuit.
 5. The adaptive noise cancellationmicrophone system according to claim 4, wherein the adaptive noisecancellation controller comprises a power estimator unit being adaptedto receive each of the speech enhanced signal, the speech nulled signaland the output filtered signal, and to generate a corresponding powersignal to be received by a silence detector unit and a signal detectorunit; the silence detector unit for detecting whether an acousticalsignal is present in the speech enhanced signal input, the speech nulledsignal and the output filtered signal, and being adapted to generate afirst output signal which indicates whether the acoustical signal ispresent; a signal detector unit for detecting whether a desired signalis present in the speech enhanced signal, the speech nulled signal andthe output filtered signal, and being adapted to generate a secondoutput signal which indicates whether the desired signal is present; anadaptive noise cancellation filter controller unit being adapted toreceive the first output signal and the second output signal todetermine the characteristic of the first input signal and the secondinput signal, and to generate the ANCF control signal which representsan updated coefficient parameter for updating the coefficients of the atleast one filter of the adaptive noise cancellation filtering circuit;and an SFS controller unit being adapted to receive the first outputsignal and the second output signal to determine the characteristic ofthe first input signal and the second input signal, and to generate theSFS control signal which represents an updated coefficient parameter forupdating the coefficients of the at least one filter of the SFS.
 6. Theadaptive noise cancellation microphone system according to claim 5,wherein the adaptive noise cancellation controller further comprises apost processing filter controller unit, wherein the post processingfilter controller unit is adapted to receive the first output signal andthe second output signal to determine the characteristic of the firstinput signal and the second input signal, and to generate a postprocessing control signal which represents an updated coefficientparameter for updating the coefficients of at least one filter of thepost processing filtering circuit.
 7. The adaptive noise cancellingmicrophone system according to claim 3, wherein the SFS comprises afirst FIR filter and a second FIR filter for receiving the first inputsignal, wherein the first FIR filter and the second FIR filter areadapted to determine the acoustical signal component in the first inputsignal, and to generate a first FIR filtered signal and a second FIRfiltered signal, respectively, representing the acoustical signalcomponent in the first input signal; a first adder for adding the firstFIR filtered signal to the second input signal, thereby generating thespeech enhanced signal; a second adder or subtracting the second FIRfiltered signal from the second input signal, thereby generating thespeech nulled signal.
 8. The adaptive noise canceling microphone systemaccording to claim 7, wherein the SFS further comprises a Least MeansSquared (LMS) algorithm block, wherein the LMS algorithm block isadapted to receive the first input signal, the SFS control signal andthe speech nulled signal, and to update the coefficients of the firstand/or second FIR filters.
 9. The adaptive noise canceling microphonesystem according to claim 7, wherein the SFS further comprises a firstRead Only Memory (ROM) and a second Read Only Memory (ROM), whereininformation representing the coefficients of the first FIR filter andthe second FIR filter are stored in the first ROM and the second ROM,and the stored information in the ROMs are used as coefficients for thefirst FIR filter and the second FIR filter, respectively.
 10. Theadaptive noise canceling microphone system according to claim 3, whereinthe adaptive noise cancellation filtering circuit comprises: an adaptivefilter being adapted to receive the speech nulled signal, and togenerate a filtered signal representing noise in the first and secondinput signal; and an adder for subtracting the filtered signal from thespeech enhanced signal, and generating an error signal; wherein the ANCFcontrol signal is used to update the coefficients of the adaptivefilter, and wherein the error signal from the adder is used as feedbackinformation to the adaptive filer.
 11. The adaptive noise cancellationmicrophone system according to claim 10, further comprising a delaycircuit arranged between the speech enhanced signal and the adder,wherein the delay circuit is adapted to receive and delay the speechenhanced signal, and to generate a delayed signal to the adder.
 12. Theadaptive noise cancellation microphone system according to claim 11,further comprising an additional, adder arranged between the speechnulled signal and the adaptive filter for subtracting an additionalfiltered signal of an additional adaptive filter from the speech nulledsignal, and generating an additional error signal to the adaptivefilter; and the additional adaptive filter being adapted to receive theerror signal, and to generate the additional filtered signalrepresenting an estimated component of the desired signal in the errorsignal, wherein the ANCF control signal is used to update thecoefficients of the additional adaptive filter, and wherein theadditional error signal is used as a feedback information to theadditional adaptive filter.
 13. The adaptive noise cancellationmicrophone system according to claim 10, further comprising anadditional adder arranged between the speech nulled signal and theadaptive filter for subtracting an additional filtered signal of anadditional adaptive filter from the speech nulled signal, and generatingan additional error signal to the adaptive filter; and the additionaladaptive filter being adapted to receive the error signal, and togenerate the additional filtered signal representing an estimatedcomponent of the desired signal in the error signal, wherein the ANCFcontrol signal is used to update the coefficients of the additionaladaptive filter, and wherein the additional error signal is used as afeedback information to the additional adaptive filter.
 14. The adaptivenoise cancellation microphone system according to claim 3, wherein theadaptive noise cancellation controller comprises a power estimator unitbeing adapted to receive each of the speech enhanced signal, the speechnulled signal and the output filtered signal, and to generate acorresponding power signal to be received by a silence detector unit anda signal detector unit; the silence detector unit for detecting whetheran acoustical signal is present in the speech enhanced signal input, thespeech nulled signal and the output filtered signal, and being adaptedto generate a first output signal which indicates whether the acousticalsignal is present; a signal detector unit for detecting whether adesired signal is present in the speech enhanced signal, the speechnulled signal and the output filtered signal, and being adapted togenerate a second output signal which indicates whether the desiredsignal is present; an adaptive noise cancellation filter controller unitbeing adapted to receive the first output signal and the second outputsignal to determine the characteristic of the first input signal and thesecond input signal, and to generate the ANCF control signal whichrepresents an updated coefficient parameter for updating thecoefficients of the at least one filter of the adaptive noisecancellation filtering circuit; and an SFS controller unit being adaptedto receive the first output signal and the second output signal todetermine the characteristic of the first input signal and the secondinput signal, and to generate the SFS control signal which represents anupdated coefficient parameter for updating the coefficients of the atleast one filter of the SFS.
 15. The adaptive noise cancellationmicrophone system according to claim 2, wherein the post processingfiltering circuit comprises a FIR filter being adapted to receive theoutput filtered signal and to generate the digital result signal, acoefficient calculation block being adapted to receive the speechenhanced signal, the speech nulled signal and the output filteredsignal, and to generate a coefficient signal representing thecoefficients of the FIR filter.
 16. The adaptive noise cancellationmicrophone system according to claim 15, wherein the coefficientcalculation block is further adapted to receive a post processingcontrol signal, wherein the post processing control signal is also usedto generate the coefficient signal representing the coefficients of theFIR filler.
 17. The adaptive noise cancellation microphone systemaccording to claim 2, further comprising a Digital-to-Analog converterfor converting the digital result signal to an analog signal, therebygenerating an output result signal.
 18. The adaptive noise cancelingmicrophone system according to claim 1, further comprises a firstamplifier and a second amplifier for receiving and amplifying signalsresulting from the sounds received by the two microphones; a first lowpass filter and a second low pass filter for receiving and filtering theamplified signals from the first amplifier and the second amplifier,thereby generating analog amplified and filtered signals; a firstAnalog-to-Digital converter and a second Analog-to-Digital converter forreceiving and converting the analog amplified and filtered signals todigital signals, thereby generating the first input signal and thesecond input signal.
 19. The adaptive noise canceling microphone systemaccording to claim 1, further comprising an adaptive noise cancellationmicrophone (ANCM) controller for receiving the speech enhanced signal,the speech nulled signal and is the output filtered signal, the adaptivenoise cancellation microphone (ANCM) controller being adapted togenerate an adaptive noise cancellation filter (ANCF) control signal andan SFS control signal for updating coefficients of at least one filterof the SFS and adaptive noise cancellation filtering circuit,respectively, using the received speech enhanced signal, the speechnulled signal and the output filtered signal.